Re: [OpenSIPS-Users] Getting the $rb or media port for egress message.

[Pavel Eremin]

Hi, if I understand correctly, you have to get $rb from onreply_route.

ср, 27 мар. 2024 г. в 02:19, Matthew Schumacher :
>
> Hello,
>
> I'm trying to log the media port for another system and using this code
> to grab the m line out of an SDP header:
>
>if (has_body("application/sdp")){
>  rtpengine_offer();
>  $var(mline) = $(rb{sdp.line,m,0});
>  xlog("TEST $var(mline)\n");
>}
>
> The problem is that it only sees the ports for ingress SDP messages and
> not egress SDP messages.  What can I do to grab the SDP messages leaving
> opensips?
>
> I want to take this data and inject a firewall rule that allows the
> media and remove it later because NAT helpers don't work on TLS traffic,
> my system already knows what port to expect, and I don't like having a
> huge pile of UDP ports allowed through.
>
> Honestly, I'm not sure why the kernel mode forwarding in rtpengine
> doesn't also allow traffic based on ports we are expecting to see
> traffic from that's what I thought it did at first until I realized
> that it's only forwarding the RTP packet through the kernel faster
> without needing userspace.
>
> Thanks,
> Matt
>
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Re: [OpenSIPS-Users] Openisps 3.2 is crashing randomly .

[Pavel Eremin]

Hi
1. Try to run with default config - it will give you a way to understand if
crash depends on configuration
If it will start successfully then just check config again params specific
for new VM.

пт, 10 февр. 2023 г. в 15:45, Sasmita Panda :

> Hi All ,
>
> I have been using opensips 3.2 for the last 1 year . But it suddenly
> crashed when I created a new machine from an image and started opensips .
>
> opensips -V
>
>
>
>
>
> *version: opensips 3.2.3 (x86_64/linux)flags: STATS: On, DISABLE_NAGLE,
> USE_MCAST, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, HP_MALLOC, DBG_MALLOC,
> FAST_LOCK-ADAPTIVE_WAITADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE
> 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535poll method
> support: poll, epoll, sigio_rt, select.svn revision: 3831:3864main.c
> compiled on 07:29:49 Jun 21 2022 with gcc 10*
>
> *https://pastebin.com/4RQNCzGW *
>
> Can anyone help me with this please ?
>
> *Thanks & Regards*
> *Sasmita Panda*
> *Senior Network Testing and Software Engineer*
> *3CLogic , ph:07827611765*
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Re: [OpenSIPS-Users] Statistics module

[Pavel Eremin]

Thanks! Yes I suppose it's useful in case of recognizing which IP or user
is most active. For example if i have opened server to whole word, then i
can do this by collecting avg_1m stats and make then live for 2mins, so,
when i ask stat by prometheus server, opensips will send limited data. In
current case opensips will send as much statistics as collected. Of course
after some time shared memory are end.

вт, 28 июн. 2022 г. в 13:55, Răzvan Crainea :

> Hi, Pavel!
>
> No, there is currently no way to set a lifetime, it will live forever.
> Please open a feature request[1] if you find this feature useful.
>
> [1] https://github.com/OpenSIPS/opensips/issues
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Core Developer
> http://www.opensips-solutions.com
>
> On 6/28/22 11:03, Pavel Eremin wrote:
> > Hi, all, does anyone work with |stat_series_profile, it seems very
> useful.|
> >
> > My question is if some value was created by update_stat_series, then
> > this stat variable will lives forever, even it 0.
> >
> > Is it possible to set the lifetime for series?
> >
> >
> >
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> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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[OpenSIPS-Users] Statistics module

[Pavel Eremin]

Hi, all, does anyone work with stat_series_profile, it seems very useful.

My question is if some value was created by update_stat_series, then this
stat variable will lives forever, even it 0.

Is it possible to set the lifetime for series?
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Re: [OpenSIPS-Users] SBC

[Pavel Eremin]

Hey!
In short, you should use the uac_auth() function in failure_route if you
are using default opensips.cfg.
If the carrier(trunk) wants you to be REGISTERED, then you have to use
restaurant module also.



вс, 9 янв. 2022 г. в 19:35, Gokan Atmaca :

> Hello
>
> (I sent wrong email before.) I installed Opensip. I created users and
> registered with zoiper. Users can call each other.
> I want to do sip authentication (trunk) for outgoing calls.
> Can you help me with this?
>
> On Sun, Jan 9, 2022 at 5:28 PM Gokan Atmaca  wrote:
> >
> > Hello
> >
> > I have installed Opensip. I can talk to internals. Trunk I want to
> > burn. I have the SIP username and password. How can I enter this
> > information? How can I make a trunk?
> >
> > Thanks.
> >
> >
> > --
> > ⢀⣴⠾⠻⢶⣦⠀
> > ⣾⠁⢠⠒⠀⣿⡁ Debian - The universal operating system
> > ⢿⡄⠘⠷⠚⠋⠀ https://www.debian.org
> > ⠈⠳⣄
>
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Re: [OpenSIPS-Users] mid_registrar errors

[Pavel Eremin]

Thanks for your answer I suppose there is some little mistake in the code
of mid_registrar due to this error easylly reproducing on the new opensips
installation.
Conditions are simple: centos 7 or debian 10, opensips 3.2 ver, use
mid_registrar (mode=2 or 1) and registrar in the same script.
For example for REGISTER with tU like 1000 we will make mid_registrar_save
and for 2 we will do save().
In case of save() we will get those errors in logs.

I have created an issue(https://github.com/OpenSIPS/opensips/issues/2716)
on github about that.



пн, 3 янв. 2022 г. в 21:34, Bogdan-Andrei Iancu :

> Hi Alexey,
>
> What the errors say is that mid-registrar, upon deleting/expiring a
> registered AOR, it is not able to push forward the un-register to the main
> registrar. This happens as some internal key (attached to the AOR) is not
> found by the mid-registrar.
>
> Not sure how this may be related to the changing of the IPs, but I guess
> this issue went away after some time, right? if not, you may stop opensips
> mid-registrar, purge the "location" table (or how you named it) and start
> opensips back. This will flush all the pending (potential bogus records)
> and give you a fresh start. Note that your device will have to re-register
> in order to have the calling working again.
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>   https://www.opensips-solutions.com
> OpenSIPS eBootcamp 2021
>   https://opensips.org/training/OpenSIPS_eBootcamp_2021/
>
> On 12/27/21 10:15 AM, Alexey Kazantsev via Users wrote:
>
> Hi list.
>
> What do these errors mean?
> This started after migrating the virtual machine and changing IP addresses.
> Everything else remain unchanged.
>
> Version 3.2.2
>
> ERROR:mid_registrar:unregister_record: 'from' key not found, skipping
> De-REGISTER
> ERROR:mid_registrar:mid_reg_aor_event: failed to unregister contact
>
>
> ---
> BR, Alexey
> https://alexeyka.zantsev.com/
>
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>
>
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Re: [OpenSIPS-Users] How to get all listening socket IP address as variable

[Pavel Eremin]

I am not sure, but seems you are sending calls to the same IP that you have
on Opensips? like SIP -> IP_A (opensips) -> IP_B(OPensips).
if you want to send a call through an external interface just send call to
some external IP (like sethost("ANY EXTERNAL IP");) and opensips will do
that with an interface like unix do.


socket=udp:enp1s0:5060   # INTERNAL_INTERFACE (192.168.1.2)

socket=udp:enp2s0:5060   # EXTERNAL_INTERFACE (DHCP)

   .

   .

   # From Internal to External

   # Testing one number only first 634 to 605

  if ($rU=~"^605") {

  sethostport("EXTERNAL_IP:5060");

   $socket_out = "udp:EXTERNAL_IP:5060";

...



   # From External to internal

   # Testing one number only first 605 to 634

if ($rU=~"^634"){

  sethostport("192.168.1.2:5060");

   $socket_out = "udp:192.168.1.2:5060";

...

чт, 18 нояб. 2021 г. в 07:53, Muhamad Putra Abdullah :

> Hi,
>
>
>
> Ok. I’m totally in learning process.
>
> Let me explain my test deployment. I have 2 separate network that
> connected via opensips server.
>
>
>
> Internal SIP Server   Opensips Server
> (2 NIC) External SIP Server
>
> (User 634) 192.168.1.1  - - - - - - - -  192.168.1.2|   (EXTERNAL_IP)
> DHCP IP - - - - - - - - 172.16.16.1 (User 605)
>
>
>
> What my intension is to relay calls between Internal and External SIP
> server via Opensips server.
>
> I use this method to achieve this:
>
>
>
>socket=udp:enp1s0:5060   # INTERNAL_INTERFACE (192.168.1.2)
>
> socket=udp:enp2s0:5060   # EXTERNAL_INTERFACE (DHCP)
>
>.
>
>.
>
># From Internal to External
>
># Testing one number only first 634 to 605
>
>   if ($rU=~"^605") {
>
>   sethostport("EXTERNAL_IP:5060");
>
>$socket_out = "udp:EXTERNAL_IP:5060";
>
>rtpproxy_engage("ier");
>
>route(relay);
>
>exit;
>
>}
>
>
>
># From External to internal
>
># Testing one number only first 605 to 634
>
> if ($rU=~"^634"){
>
>   sethostport("192.168.1.2:5060");
>
>$socket_out = "udp:192.168.1.2:5060";
>
>rtpproxy_engage("eir");
>
>route(relay);
>
>exit;
>
>}
>
>
>
> I have no problem on call from external to internal because I know the IP
> address of internal opensips server, the problem is call from internal to
> external.
>
> Is there better solutions than this?
>
>
>
> Regards
>
>
>
> *From:* Bogdan-Andrei Iancu 
> *Sent:* Wednesday, 17 November, 2021 8:24 PM
> *To:* Muhamad Putra Abdullah ; OpenSIPS users
> mailling list 
> *Subject:* Re: [OpenSIPS-Users] How to get all listening socket IP
> address as variable
>
>
>
> If so, when you receive a request on the external IP, the $socket_in(ip)
> should return the actual DHCP IP.
>
> Regards,
>
> Bogdan-Andrei Iancu
>
>
>
> OpenSIPS Founder and Developer
>
>   https://www.opensips-solutions.com
>
> OpenSIPS eBootcamp 2021
>
>   https://opensips.org/training/OpenSIPS_eBootcamp_2021/
>
> On 11/17/21 10:21 AM, Muhamad Putra Abdullah wrote:
>
> Hi,
>
>
>
> socket=udp:enp1s0:5060 tag INTERNAL_IP  # CUSTOMIZE ME
>
> socket=udp:enp2s0:5060 tag EXTERNAL_IP  # CUSTOMIZE ME
>
>
>
> My INTERNAL_IP is set to static and EXTERNAL_IP set to DHCP. I just use
> tag to test something.
>
>
>
> Regards
>
>
>
> *From:* Bogdan-Andrei Iancu  
> *Sent:* Wednesday, 17 November, 2021 3:55 PM
> *To:* Muhamad Putra Abdullah 
> ; OpenSIPS users mailling list
>  
> *Subject:* Re: [OpenSIPS-Users] How to get all listening socket IP
> address as variable
>
>
>
> Hi,
>
> How do you define the listening socket in OpenSIPs cfg ? (the one related
> to the DHCP interface)
>
> Regards,
>
>
> Bogdan-Andrei Iancu
>
>
>
> OpenSIPS Founder and Developer
>
>   https://www.opensips-solutions.com
>
> OpenSIPS eBootcamp 2021
>
>   https://opensips.org/training/OpenSIPS_eBootcamp_2021/
>
> On 11/17/21 2:19 AM, Muhamad Putra Abdullah wrote:
>
> Hi,
>
>
>
> Is there a way to get the IP address of DHCP interface to use in the
> script? I can get the call go through if I set both the interface as static.
>
>
>
> Regards
>
>
>
> Get Outlook for Android 
>
>
> --
>
> *From:* Bogdan-Andrei Iancu  
> *Sent:* Tuesday, November 16, 2021, 6:45 PM
> *To:* OpenSIPS users mailling list; Muhamad Putra Abdullah
> *Subject:* Re: [OpenSIPS-Users] How to get all listening socket IP
> address as variable
>
>
> Hi,
>
> via the socket_xx() 

Re: [OpenSIPS-Users] How to get all listening socket IP address as variable

[Pavel Eremin]

Hey,
to relay with specific interface i suppose you have use function
force_send_socket();
i am sure only about that scenario:
"
socket=udp:eth0:5060 as 1.1.1.1  #dhcp interface with external ip
socket=udp:2.2.2.2:5061
...
force_send_socket("udp:1.1.1.1:5060");
t_relay();
""

вт, 16 нояб. 2021 г. в 05:51, Muhamad Putra Abdullah :

> Hi,
>
>
>
> The reason I’m asking is because one of the interface need to be in DHCP
>  so I need the IP address information for serthostport() and socket_out()
> to relay SIP/RTP to other interface.
>
>
>
> Thanks
>
>
>
> *From:* Muhamad Putra Abdullah
> *Sent:* Monday, 15 November, 2021 3:39 PM
> *To:* users@lists.opensips.org
> *Subject:* How to get all listening socket IP address as variable
>
>
>
> Hi,
>
>
>
> I have 2 listening interface for opensips 3.2. How do I get both IP
> address to be used as variable in opensips config file? I try to use
> socket_in/ socket_out but failed to get the other interface IP address.
>
>
>
> Thanks
>
>
>
> Sent from Mail  for
> Windows
>
>
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Re: [OpenSIPS-Users] opensips control panel error connecting to rules, carrier group

[Pavel Eremin]

HI, add  require("../../../common/cfg_comm.php");
before "require("template/header.php");"

works for me. Maybe a little bug.

чт, 21 окт. 2021 г. в 22:12, Eamon Garland :

> Hi Bogdan
> i am using the following for the install
>
> wget https://github.com/OpenSIPS/opensips-cp/archive/master.zip
>
> Regards,
> Eamon
> On Thu, Oct 21, 2021 at 5:55 PM Bogdan-Andrei Iancu 
> wrote:
>
>> Hi Eamon,
>>
>> What version of CP do you use ?
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>>
>> OpenSIPS Founder and Developer
>>   https://www.opensips-solutions.com
>> OpenSIPS eBootcamp 2021
>>   https://opensips.org/training/OpenSIPS_eBootcamp_2021/
>>
>> On 10/20/21 8:27 PM, Eamon Garland wrote:
>>
>> Hi
>> i have installed latest version on debian 10 and get the following error
>> when i try to add rules
>>
>>  PHP Fatal error:  Uncaught Error: Call t*o undefined function
>> get_group()* in
>> /var/www/html/opensips-cp/web/tools/system/drouting/template/header.php:33\nStack
>> trace:\n#0
>> /var/www/html/opensips-cp/web/tools/system/drouting/rules.php(24):
>> require()\n#1 {main}\n  thrown in
>> /var/www/html/opensips-cp/web/tools/system/drouting/template/header.php on
>> line 33, referer: http://x.x.x.x/cp/tools/system/drouting/gateways.php
>>
>> [image: cdca5e74-523d-4cc2-b56b-f598aed8535d]
>>
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>>
>>
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Re: [OpenSIPS-Users] Fwd: Retransmission and 300 Redirect

[Pavel Eremin]

Many thanks for your advices.

Чт, 22 окт. 2020 г. в 18:44, Răzvan Crainea :

> Hi, Pavel!
>
> I think you can play with the T1 and T2 timers[1] to avoid, or at least
> control better the re transmissions, however note that these settings
> affect all transactions, not just the INVITE transaction one you're
> targeting. TBH, I wouldn't do that.
>
> Another possibility would be to completely avoid re-transmissions is to
> go stateless (instead of using t_relay(), just use send() and the
> generic onreply_route).
>
> However, the best way to achieve this, is to fix the redirect_server to
> be able to either send a 100 Trying within a reasonable time, or at
> least be able to absorb re-transmissions.
>
> In order to make sure you're only processing a single 300 Redirect, you
> can use a flag that says whether the redirect was processed or not, and
> only reprocess it if the flag is not set.
>
> [1] https://opensips.org/docs/modules/3.1.x/tm.html#param_T1_timer
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Core Developer
> http://www.opensips-solutions.com
>
> On 10/22/20 12:32 PM, Pavel Eremin wrote:
> >
> > Hi, community!
> > Is anyone can help me to understand how to avoid 2 situations:
> >
> > 1. I don't want to send retransmission  (INVITE) from "opensips" to
> > "redirect" server, even "redirect" does not answer.
> >
> > 2. If I have sent a few invites to "redirect" , then i will receive a
> > few 300 Redirect messages, and "opensips" will generate branches for all
> > contacts in all "300 Redirect" messages. how can i ignore all "300
> > Redirects".
> >
> > I am not sure that t_check_trans() can help me with that.
> > Thanks. I added a picture with the call flow I have.
> >
> > ___
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> >
>
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[OpenSIPS-Users] Fwd: Retransmission and 300 Redirect

[Pavel Eremin]

Hi, community!
Is anyone can help me to understand how to avoid 2 situations:

1. I don't want to send retransmission  (INVITE) from "opensips" to
"redirect" server, even "redirect" does not answer.

2. If I have sent a few invites to "redirect" , then i will receive a few
300 Redirect messages, and "opensips" will generate branches for all
contacts in all "300 Redirect" messages. how can i ignore all "300
Redirects".

I am not sure that t_check_trans() can help me with that.
Thanks. I added a picture with the call flow I have.
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Re: [OpenSIPS-Users] opensips as load-balancer - configuration

[Pavel Eremin]

Hello, my 5 cents for understanding load balancer scheme:

https://opensips.org/pub/events/2016-05-10_OpenSIPS-Summit_Amsterdam/Giovanni_Maruzzelli-OpenSIPS_Summit2016-FreeSWITCH_HA.pdf

ср, 18 дек. 2019 г. в 22:31, Abu Ibrahim via Users :

> Hi,
>
> I recently started using opensips and installed 3.0 version by following
> the steps available at opensips.org. My requirement is just to set up
> Opensips as load-balancer between two backend sip servers (actually two
> freeswitch servers) which will be working as registrars and call routing
> servers. So my opensips will just be load balancing REGISTER and INVITEs to
> my two backend freeswitch servers.
>
> After installing the opensips, I've tried route scripts mentioned on the
> following two links but none of them is working and calls are always failed
> with 404 not found.
>
> https://www.opensips.org/Documentation/Tutorials-LoadBalancing-1-9
>
> https://freeswitch.org/confluence/display/FREESWITCH/Enterprise+deployment+OpenSIPS
>
> I am attaching my configuration details in the text file here.
>
> Can someone please help me further what exact or minimal configuration and
> routing script required at opensips side to accept my traffic and load
> balance as mentioned above? Also, can someone please let me know a good
> tutorial/link where basic call routing, endpoints definition or related
> information for opensips are explained and will be helpful for me to start
> off with the call testing. Thanks in advance
>
> Regards
>
>
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Re: [OpenSIPS-Users] Registration permissions per username

[Pavel Eremin]

Hello why not regexp not working for you. I give you my example that works
fine.(but with 1.11)
to reload i use command opensipsctl fifo regex_reload

[config]
 REGEX
loadmodule "regex.so"
modparam("regex", "file", "/usr/out_isp/etc/opensips/regex_groups")

...
if (pcre_match_group("$fU", "0")) {
}
...

[content regex_group file]
[0]

^1000
^1001
^1003
^50065
^anyname_from_start
anyname_in_anywhere of $fU




вт, 2 июл. 2019 г. в 10:33, Alexey Kazantsev via Users <
users@lists.opensips.org>:

> Hi list,
>
> is it possible to filter REGISTER requests with permissions.so [1] module,
> based on username?
>
> It's written " Main purpose of the function is to prevent registration of
> "prohibited" IP addresses. " When speaking about IP filtering,
> I'd rather use check_address or check_source_address functions.
>
> But now I'd like to filter by userame, because users may register from
> random
> addresses.
>
> I tried to create pairs of regexps in register.allow and register.deny
> files,
> but no success. Maybe I've done something wrong.
>
>
> [1]
> https://opensips.org/html/docs/modules/3.0.x/permissions.html#sec-registration-permissions
>
> ---
> BR, Alexey
> http://alexeyka.zantsev.com/
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Re: [OpenSIPS-Users] opensips transparent technology

[Pavel Eremin]

$ru and $rU are something like destination number.
Carrier will show $fu (if it according carrier's rules). Carriers ignore
domain in $fu and show "display" or user from $fu.

2016-02-02 9:33 GMT+05:00 MichaelLeung <gbcbook...@gmail.com>:

> yes , $ru
>
> does the carrier will display the "ringing from Name " base on the string
> $ru ? i doubt it b because string $ru include domain of sip server string,
> or string $rU  will also be sent into carrier and will be accepted as the
> display number when ringing the phone only when we change it to a real
> phone number.
>
>
> On 02/01/2016 05:50 PM, Bogdan-Andrei Iancu wrote:
>
> Hi,
>
> Assuming you were talking about $ru (and not $ur), the answer is yes, the
> new $ru will be set into the SIP request and sent out to the next SIP hop
> (your carrier). Note that $ru will push changes only in the Request URI.
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 01.02.2016 10:44, MichaelLeung wrote:
>
> does $ur willl be transmitted to carriar when i change the uri to a real
> number  ?
>
> On 01/29/2016 01:35 PM, MichaelLeung wrote:
>
> ok, thanks .
>
>
> On 01/29/2016 12:05 PM, Pavel Eremin wrote:
>
> Opensips will not change CLI or CAllerID (real number from carrier) if you
> don't tell him to do it. So, It's like ask what name for "painting white
> pages" - there is no name it's just blank pages.. I think
>
>
> 2016-01-29 8:08 GMT+05:00 MichaelLeung < <gbcbook...@gmail.com>
> gbcbook...@gmail.com>:
>
>> any reply ?
>>
>>
>> On 01/26/2016 04:42 PM, MichaelLeung wrote:
>>
>> can uac_replace_from read real phone number from databases?
>>
>> On 01/25/2016 01:03 PM, MichaelLeung wrote:
>>
>> thanks for reply
>> no , it is just a asking , i don't have real phone number database, or
>> should i have one ?
>> can you tell me what is the name of this technology ?
>>
>> On 01/24/2016 07:33 PM, Stefano Pisani wrote:
>>
>> Where is their real phone number?
>> Do you have it in a database?
>> You can change the From header to show the real phone number.
>>
>>
>>
>> Il 24/01/2016 12.22, MichaelLeung ha scritto:
>>
>> Hi all
>>
>> i was trying to make my opensips users to sent their real phone number
>> when they call .
>>
>> what is the name of this technology ? transmit transparently ?
>>
>> i search google find nothing, and where can i read document of this
>> technology ?
>>
>> thanks.
>>
>>
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Re: [OpenSIPS-Users] opensips transparent technology

[Pavel Eremin]

Opensips will not change CLI or CAllerID (real number from carrier) if you
don't tell him to do it. So, It's like ask what name for "painting white
pages" - there is no name it's just blank pages.. I think


2016-01-29 8:08 GMT+05:00 MichaelLeung :

> any reply ?
>
>
> On 01/26/2016 04:42 PM, MichaelLeung wrote:
>
> can uac_replace_from read real phone number from databases?
>
> On 01/25/2016 01:03 PM, MichaelLeung wrote:
>
> thanks for reply
> no , it is just a asking , i don't have real phone number database, or
> should i have one ?
> can you tell me what is the name of this technology ?
>
> On 01/24/2016 07:33 PM, Stefano Pisani wrote:
>
> Where is their real phone number?
> Do you have it in a database?
> You can change the From header to show the real phone number.
>
>
>
> Il 24/01/2016 12.22, MichaelLeung ha scritto:
>
> Hi all
>
> i was trying to make my opensips users to sent their real phone number
> when they call .
>
> what is the name of this technology ? transmit transparently ?
>
> i search google find nothing, and where can i read document of this
> technology ?
>
> thanks.
>
>
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[OpenSIPS-Users] cdrtool and toll-free numbers

[Pavel Eremin]

Hi, all! can CDRTool rate ordinary call and call to Toll-Free numbers at
the same time?

I know that we can change field Username (BillingPartyId) but how to use
field FROM  for Toll-Free numbers and Username for ordinary calls.
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Re: [OpenSIPS-Users] Fraud Detection in OpenSIPS 1.12

[Pavel Eremin]

It's only discussion about new module, no real module available.
07.11.2014 11:29 пользователь Лытаев Антон Викторович l...@ptcomm.ru
написал:

 This unit is compatible with version Opensips 1.11.2-tls (x86_64)? You can
 get a direct link?

 02.09.2014 20:26, Răzvan Crainea wrote:

 Hi all,

 The second topic discussed during the last IRC meeting[1] was about
 building a Fraud Detection module that prevents PBX or accounts hijacking.
 ..




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Re: [OpenSIPS-Users] [Paid support needed] Fix OpenSIPS configuration to add TCP support

[Pavel Eremin]

Already make a bid as Erewin.
05.10.2014 20:36 пользователь Adam Raszynski netcentr...@gmail.com
написал:

 Hi All,

 I'm looking for paid support in fixing my opensips config file

 All interested OpenSIPS hackers are welcome:


 https://www.freelancer.pl/projects/Software-Architecture-Linux/Fix-OpenSIPS-configuration-add-TCP.html

 Hope that's good group, I've searched but didn't find better place to post

 Regards

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Re: [OpenSIPS-Users] CDRtool

[Pavel Eremin]

Thanks, i read all of docs and have question about performance. (I Did
describe it in another email to community)
30.09.2014 4:56 пользователь Adrian Georgescu a...@ag-projects.com
написал:

 If you want to dive this deep into CDRTool, it would be a good start to
 read the documentation, that is all *.txt files in docs/ folder.

 --
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[OpenSIPS-Users] CDRtool and perfomance with short calls( 3 sec )

[Pavel Eremin]

Hi, All! I am using CDRTool with prepaid and when reached 15 calls per
second i have  low performance.

As I understood CDRtool engine gets ALL active session and recalc
MaxSessionTime for ALL calls when new call happens?

so if i have many short calls CDRTool has heavy load.

I don't want to recalc active session. (let negative balance be).

Can i do it without make a lot of changes in code? may be disable some
function in rating.php?

PS: I have read docs about PREPAID This scheme provides a
fair balancing policy with small performance penalty on the servers.
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[OpenSIPS-Users] callcontrol 2.1.0

[Pavel Eremin]

Hi, All!
Trying to use callcontrol 2.1.0 and always get prepaid = None.
I can't see that callcontrol send request to CDRTool Engine? There is my
logs:

Sep 29 11:49:39 fiber4 call-control[15122]: Got request: init: callid=
303-7439@103.239.147.16 from=pavel sip:pa...@sip.fiberpipe.in:5060 ruri=
sip:7922633004@103.239.147.14:5060 diverter=None
sourceip=103.239.147.16 prepaid=None call_limit=None

Sep 29 11:49:39 fiber4 call-control[15122]: Call id 303-7439@103.239.147.16
added to list of controlled calls
*!!! I think here callcontrol must ask CDRTool but it did not!!! *
Sep 29 11:49:39 fiber4 call-control[15122]: Call id 303-7439@103.239.147.16
of pa...@sip.fiberpipe.in to sip:7922633004@103.239.147.14:5060 is postpaid
not limited

Sep 29 11:49:39 fiber4 call-control[15122]: Call id 303-7439@103.239.147.16
removed from the list of controlled calls

Sep 29 11:49:39 fiber4 call-control[15122]: Sent reply: No limit
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Re: [OpenSIPS-Users] callcontrol 2.1.0

[Pavel Eremin]

Thanks a lot! It's so simple)



2014-09-29 18:13 GMT+06:00 Venkatesh Macha linuxven...@gmail.com:

 Hi pavel,

   Are you setting prepaid_account_flag in opensips.cfg, i think
 callcontrol is always treating your call as postpaid. so try to set
 prepaid_account_flag.

 Venkatesh Macha,
 Junior VOIP Engineer,
 @ sillycodes http://sillycodes.com




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[OpenSIPS-Users] CDRtool

[Pavel Eremin]

Hi, All! Let's dance with CDRTool

I am trying to use it on heavy load system and get stack.

Why in this code author match canonical number with string started by 0?(in
hard way) Because of this line all calls are postpaid...:(

   * if
(!preg_match(/^0[9-0]{1,}@/,$CDR-CanonicalURINormalized))* {
$log=sprintf (MaxSessionTime=unlimited Type=prepaid
CallId=%s BillingParty=%s
DestId=None,$NetFields['callid'],$CDR-BillingPartyId);
syslog(LOG_NOTICE, $log);
$this-logRuntime();
$ret=none.\n.type=prepaid;
return $ret;
} else {
if (!$CDR-DestinationId) {
$log = sprintf (error: cannot figure out the
destination id for %s,$CDR-CanonicalURI);
$this-logRuntime();
syslog(LOG_NOTICE, $log);
$ret=$log.\n.type=prepaid;
return $ret;
}
}
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Re: [OpenSIPS-Users] CDRtool

[Pavel Eremin]

This code in rating.php...

2014-09-29 22:18 GMT+06:00 Pavel Eremin eremina@gmail.com:

 Hi, All! Let's dance with CDRTool

 I am trying to use it on heavy load system and get stack.

 Why in this code author match canonical number with string started by
 0?(in hard way) Because of this line all calls are postpaid...:(

* if
 (!preg_match(/^0[9-0]{1,}@/,$CDR-CanonicalURINormalized))* {
 $log=sprintf (MaxSessionTime=unlimited Type=prepaid
 CallId=%s BillingParty=%s
 DestId=None,$NetFields['callid'],$CDR-BillingPartyId);
 syslog(LOG_NOTICE, $log);
 $this-logRuntime();
 $ret=none.\n.type=prepaid;
 return $ret;
 } else {
 if (!$CDR-DestinationId) {
 $log = sprintf (error: cannot figure out the
 destination id for %s,$CDR-CanonicalURI);
 $this-logRuntime();
 syslog(LOG_NOTICE, $log);
 $ret=$log.\n.type=prepaid;
 return $ret;
 }
 }


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[OpenSIPS-Users] CDRTool IP Auth

[Pavel Eremin]

Hi people!

We need to implement some features in CDRTool and ready to work with
developer who can do it.

Core:
1. Prepaid based on IP Authorization.

Web:
For customer:
Add some pages and other things.

1. Balance  - default page with account information,
2. Balance History - recharge history(+balance, -balance),
3. Source IP in CDR lists,
4. Rates - rate list like in Admin part of portal,

Main aim is Core part.

All things above will be, a patch to CDRtool, so i will be available for
all community.
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Re: [OpenSIPS-Users] Remove prefix from number

[Pavel Eremin]

Why not use Dialplan module?

...
dp_translate(1,$fU/$var(from_user))
uac_replace_from(,sip:$var(from_user)$fd)
...

On Mon, 29 Aug 2011 01:53:10 +0600, s...@veliko-turnovo.com wrote:


Dear all,

I try to remove prefix from phone number without success.

$avp(prefix)=159#;
$avp(number)=$fU;
avp_subst($avp(number)/$avp(ret)/g, /$avp(pref)//g);

How to do this?

Best regards,
Plamen


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Re: [OpenSIPS-Users] topology_hiding Record-Route

[Pavel Eremin]

:42:36 opensips ./opensips[20031]:  
DBG:db_postgres:db_postgres_val2str: PQescapeStringConn: in: 44 chars,  
out: 44 chars
Aug  9 09:42:36 opensips ./opensips[20031]:  
DBG:db_postgres:db_postgres_val2str: PQescapeStringConn: in: 6 chars, out:  
6 chars
Aug  9 09:42:36 opensips ./opensips[20031]:  
DBG:db_postgres:db_postgres_val2str: PQescapeStringConn: in: 3 chars, out:  
3 chars
Aug  9 09:42:36 opensips ./opensips[20031]:  
DBG:db_postgres:db_postgres_val2str: PQescapeStringConn: in: 20 chars,  
out: 20 chars
Aug  9 09:42:36 opensips ./opensips[20031]:  
DBG:db_postgres:db_postgres_val2str: PQescapeStringConn: in: 20 chars,  
out: 20 chars
Aug  9 09:42:36 opensips ./opensips[20031]:  
DBG:db_postgres:db_postgres_val2str: PQescapeStringConn: in: 3 chars, out:  
3 chars
Aug  9 09:42:36 opensips ./opensips[20031]:  
DBG:db_postgres:db_postgres_val2str: PQescapeStringConn: in: 42 chars,  
out: 42 chars
Aug  9 09:42:36 opensips ./opensips[20031]:  
DBG:db_postgres:db_postgres_val2str: PQescapeStringConn: in: 0 chars, out:  
0 chars
Aug  9 09:42:36 opensips ./opensips[20031]:  
DBG:db_postgres:db_postgres_submit_query: 0x7dadc8 PQsendQuery(insert into
sip_trace  
(msg,callid,method,status,fromip,toip,time_stamp,direction,fromtag,traced_user  
) values

('SIP/2.0 180 Ringing\\015\\012Via: SIP/2.0/UDP 11.11.11.11:5061;rport;
branch=z9hG4bK-2257328459-3759225282-335606691-1600804190\\015\\012
From:  
sip:73512453023@11.11.11.11;tag=2925795659-3759225282-335606691-1600804190\\015\\012To:  
sip:79128923945@22.22.22.22;us
er=phone;tag=1c416124843\\015\\012Call-ID:  
4b196530c23911e0a3f300145e556a5f@11.11.11.11\\015\\012CSeq: 1  
INVITE\\015\\012
Contact: sip:22.22.22.22:5060;did=1a8.a01b9366\\015\\012Supported:  
em,timer,replaces,path,early-session,resource-priority\\01
5\\012Allow:  
REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE\\015\\012
Server: Audiocodes-Sip-Gateway-/v.5.80A.039.005\\015\\012Content-Length:  
0\\015\\012
Record-Route:  
sip:33.33.33.33;lr;ftag=2925795659-3759225282-335606691-1600804190;did=1a.9b364f41\\015\\012\\015\\012',

'4b196530c23911e0a3f300145e556a5f@11.11.11.11','INVITE','180','udp:22.22.22.22:5060','udp:11.11.11.11:5061',
'2011-08-09  
09:42:36','out','2925795659-3759225282-335606691-1600804190',''))
Aug  9 09:42:36 opensips ./opensips[20031]: DBG:core:db_new_result:  
allocate 48 bytes for result set at 0x7dc168
Aug  9 09:42:36 opensips ./opensips[20031]:  
DBG:db_postgres:db_postgres_store_result: 0x7dadc8  
PQresultStatus(PGRES_COMMAND_OK) PQgetResult(0x25b3bc0)
Aug  9 09:42:36 opensips ./opensips[20031]: DBG:db_postgres:free_query:  
PQclear(0x25b3bc0) result set
Aug  9 09:42:36 opensips ./opensips[20031]: DBG:core:db_free_rows: freeing  
0 rows
Aug  9 09:42:36 opensips ./opensips[20031]: DBG:core:db_free_result:  
freeing result set at 0x7dc168
Aug  9 09:42:36 opensips ./opensips[20031]: DBG:tm:set_timer: relative  
timeout is 120
Aug  9 09:42:36 opensips ./opensips[20031]: DBG:tm:insert_timer_unsafe:  
[1]: 0x7f4a74c666d0 (87872)
Aug  9 09:42:36 opensips ./opensips[20031]: DBG:tm:t_unref: UNREF_UNSAFE:  
[0x7f4a74c66480] after is 0
Aug  9 09:42:36 opensips ./opensips[20031]: DBG:core:destroy_avp_list:  
destroying list (nil)
Aug  9 09:42:36 opensips ./opensips[20031]: DBG:core:receive_msg: cleaning  
up




On Tue, 09 Aug 2011 01:35:19 +0600, Bogdan-Andrei Iancu  
bog...@opensips.org wrote:



HI Pavel,

My first guess is that the 180 somehow does no match the  
transactionIf you can reproduce the case, could you run a call with  
debug=6 and post the opensips log ?


Regards,
Bogdan

On 08/08/2011 08:24 AM, Pavel Eremin wrote:

Hi!

iF i use topology_hiding in initial request then initial 'INVITE' are
good. There are no Record-Route or Via(except myself) header exist. But  
in

180 and 183 SIP messages Record-Route is present.

Topology_hiding must delete RR header 180 and 183 in messages too, Does  
it?


Some scheme like sip_trace. I need to hide C sip-proxy. B is
topology_hiding opensips. A - some client.

A - initial - B
C - initial - B
B - 100 try - A
C - 180 Ring - B
A - 180 Ring - B (RR present with C ip) ???
C - 200 OK - B
A - 200 OK - B  (no RR present)
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Re: [OpenSIPS-Users] Grandstream and TLS

[Pavel Eremin]

It may be helpful for other people. if you solve it, please post answer...  
i want to use TLS in near future.


On Fri, 22 Jul 2011 20:05:24 +0600, g...@dataproducts.ae wrote:




Dear Sir,

We are trying to connect a Grandstream Handy Tone 503 with Opensips 1.6.4
using TLS, and are having difficulties.
Our opensips settings are as follows:

disable_tls = no
listen = tls:x.x.x.x:300
tls_verify_server = 0
tls_verify_client = 1
tls_require_client_certificate = 0
#tls_method = TLSv1
tls_method=sslv3
tls_certificate = /usr/etc/opensips/tls/user/user-cert.pem
tls_private_key = /usr/etc/opensips/tls/user/user-privkey.pem
tls_ca_list = /usr/etc/opensips/tls/user/user-calist.pem

The error message we get on opensips is as follows:

 Jul 21 15:58:42 techdata opensips[16153]:
ERROR:core:tls_accept: some error in SSL (ret=0, err=1, errno=0/Success):
Jul 21 15:58:42 techdata opensips[16153]:
ERROR:core:tls_print_errstack: error:14094410SSL
routinesSSL3_READ_BYTESSslv3 alert handshake failure

Blink softphone works with this Opensips config in TLS mode
In ATA 503, we have SIP Transfer as TLS

Best rergards




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Re: [OpenSIPS-Users] OpenSipS 1.6 wont build with drouting module enabled

[Pavel Eremin]

On Tue, 28 Dec 2010 21:50:48 +0500, Dovid Bender os-l...@dovid.net wrote:



When trying to compile OpenSipS 1.6 trunk if the module drouting is enabled it 
just dies after compiling module domainpolicy. Any idea what would cause this ?


in my case it was because i doing svn co ... opensips_1_6 in same directory.
i have done successfully make all  when i got opensips_1_6_4 distribution and 
put it in different directory.


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Re: [OpenSIPS-Users] OpenSipS 1.6 wont build with drouting module enabled

[Pavel Eremin]

in my case it was because i doing svn co ... opensips_1_6 in same directory.
i have done successfully make all  when i got opensips_1_6_4 distribution and 
put it in different directory.


sorry, not distr, of course i got src.


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[OpenSIPS-Users] what is DROP_RATE means?...

[Pavel Eremin]

I include RATELIMIT module to my OpenSIPS installation and i have a question: 
What is DROP_RATE when i run rl_stat command...
if DROP_RATE grows is it dangerous?

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Re: [OpenSIPS-Users] T38 and changing media port for T38 SDP

[Pavel Eremin]

you may change and control content of SDP body by module textops. So, if  
port is change you may change it back or log it or smth else...
i use it when Broadworks change media port for G711 to 0...

Julian Yap julianok...@gmail.com писал(а) в своём письме Sat, 03 Jul  
2010 03:37:48 +0600:

 Any help greatly appreciated!

 I'm having problems with a T38 UA which changes port when negotiating
 T38 media.  All the other UA's I've encountered thus far use the same
 RTP port throughout.

 I'm also using RTPProxy.

 In the final 200 OK SDP, the UA changes media port from 49200 to 49152
 but this changeover isn't detected and the media is sent back to port
 49200 so the call then fails to negotiate T38 properly.  Not sure how
 to log the port changes to further debug this issue as well.

 Here is the flow:
 | UA                | OpenSIPS          | T38 GW            |
 |         INVITE SDP ( g711U)           |                   |
 |(5060)   --  (5060)   |                   |
 |         100 Trying|                   |                   |
 |(5060)   --  (5060)   |                   |
 |                   |         INVITE SDP ( g711U)           |
 |                   |(5060)   --  (5060)   |
 |                   |         100 Trying|                   |
 |                   |(5060)   --  (5060)   |
 |                   |         180 Ringing SDP ( g711U)      |
 |                   |(5060)   --  (5060)   |
 |                   |         200 OK SDP ( g711U)           |
 |                   |(5060)   --  (5060)   |
 |                   |         RTP (g711U)                   |
 |                   |(11392)  --  (14110)  |
 |         RTP (g711U)                   |                   |
 |(49200)  --  (10878)  |                   |
 |         180 Ringing SDP ( g711U)      |                   |
 |(5060)   --  (5060)   |                   |
 |         RTP (g711U)                   |                   |
 |(49200)  --  (10878)  |                   |
 |                   |         RTP (g711U)                   |
 |                   |(11392)  --  (14110)  |
 |         200 OK SDP ( g711U)           |                   |
 |(5060)   --  (5060)   |                   |
 |         ACK       |                   |                   |
 |(5060)   --  (5060)   |                   |
 |         RTP (g711U)                   |                   |
 |(49200)  --  (10878)  |                   |
 |         RTP (g711U)                   |                   |
 |(49200)  --  (10878)  |                   |
 |                   |         200 OK SDP ( g711U)           |
 |                   |(5060)   --  (5060)   |
 |                   |         RTP (g711U)                   |
 |                   |(11392)  --  (14110)  |
 |                   |         ACK       |                   |
 |                   |(5060)   --  (5060)   |
 |         200 OK SDP ( g711U)           |                   |
 |(5060)   --  (5060)   |                   |
 |         ACK       |                   |                   |
 |(5060)   --  (5060)   |                   |
 |         RTP (g711U)                   |                   |
 |(49200)  --  (10878)  |                   |
 |         RTP (g711U)                   |                   |
 |(49200)  --  (10878)  |                   |
 |                   |         ACK       |                   |
 |                   |(5060)   --  (5060)   |
 |                   |         INVITE SDP ( t38)             |
 |                   |(5060)   --  (5060)   |
 |         INVITE SDP ( t38)             |                   |
 |(5060)   --  (5060)   |                   |
 |         200 OK SDP ( t38)             |                   |
 |(5060)   --  (5060)   |                   |

 This is where it sends the 200 OK with a different media port.

 - Julian
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[OpenSIPS-Users] Unexpected loose_route() behavior

[Pavel Eremin]

I've got same trouble
and fix it in the same way...

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Re: [OpenSIPS-Users] is_from_local, missing loadmodule?

[Pavel Eremin]

Did you add Domain Module in loadmodule section?

On Mon, 07 Jun 2010 16:39:52 +0600, Premalatha Kuppan premala...@ngintech.com 
wrote:

 Hi,

 Can someone tell me, how to recover from this error,

 Jun  7 06:17:52 [10803] DBG:core:find_cmd_export_t: found sl_send_reply(2)
 in module sl [//lib64/opensips/modules/]
 Jun  7 06:17:52 [10803] DBG:core:find_cmd_export_t: is_from_local not
 found
 Jun  7 06:17:52 [10803] DBG:core:find_cmd_export_t: is_from_local not
 found
 *Jun  7 06:17:52 [10803] CRITICAL:core:yyerror: parse error in config file,
 line 351, column 22-23: unknown command is_from_local, missing loadmodule?
 *

 In my opensips.cfg file i have added the following code,

   if (!is_from_local()) {
 send_reply(403,Forbidden access to media
 service);
 exit;
 }

 hence, getting this error while staring opensips.

 Pleae help.

 Thanks,
 Prem



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Re: [OpenSIPS-Users] Unexpected loose_route() behavior

[Pavel Eremin]

I've got same trouble. And fix it like you.

On Tue, 08 Jun 2010 09:34:13 +0600, Matthew Lehner mleh...@gmail.com wrote:

 I am setting up opensips to act as a proxy between a SIP trunk
 provider and more than one asterisk server. I am using alias_db to
 determine which asterisk server a particular DID/user should be
 relayed to. I am also using record_route() to ensure my proxy stays in
 the entire dialog of the call.

 The initial requests go through just fine, but subsequent requests in
 the same dialog from the SIP provider are not getting routed properly
 because of loose_route().

 When the request from the SIP provider arrives, it hits loose_route()
 and the RURI gets changed to sip:222.222.222.227;lr=on which does not
 contain a username and so alias_db can no longer match the call
 details and route the request to the proper asterisk server.

 The way I understood loose_route() was supposed to work is.. it checks
 the top-most Route header to see if it is the local proxy.. if it is
 it removes that Route and if there is another Route below it.. it will
 change the RURI to that.

 If I just don't do loose_route() on requests from the SIP provider,
 everything works as expected.. but this does not seem like the right
 solution to the problem.

 I have included debug output from opensips, along with some of my own logging.

 333.333.333.x is the SIP provider
 222.222.222.x is my opensips proxy

 Regards,

 Matt



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